The congestion control mechanisms of Transmission Control Protocol (TCP) are very effective in providing best effort service in wired networks, where packet losses are mainly due to congestion in the network. In wireless mobile networks, more often than not, loss of packets is because of corruption of data on the wireless link. The TCP sender responds to these losses as if they are due to congestion, by reducing its congestion window, thereby reducing the rate of flow of packets. The reduction in congestion window is a necessity when network is experiencing congestion to avoid congestion collapse but it is not required if packet losses occur due to corruption of data on the wireless link. This unnecessary reduction in congestion window for corruption losses is the main reason for poor throughput of data transfer in wireless networks. The reduction in congestion window for corruption losses can be avoided if TCP can successfully differentiate between packet losses due to congestion and corruption. We suggest enhancements to TCP that, if implemented, will help the TCP receiver in separately identifying corruption losses and congestion losses. The enhancements are suggested over and above standard TCP NewReno and we call this new scheme as "NewRenoEln (NewReno with Explicit Loss Notification)". We suggest that the TCP sender attach a separate checksum for the TCP header with the packet. Since the length of the TCP header is much smaller as compared to the length of the TCP packet, there is a large probability that the TCP receiver will receive the header portion of the TCP packet without error even if the data portion of the packet is corrupted. Once the header information is found to be correct for a corrupted packet, the receiver can generate reliable Explicit 5oss Notification (ELN) for the sender. We derive an expression for the probability of a receiver generating successful Explicit Loss Notification, assuming a generic link layer protocol that is used for data transfer over wireless link. With this analysis, we show that there is large probability that receiver will generate successful ELN for various channel conditions We also suggest modifications to the sender behavior on receiving successful Explicit Loss Notification from the receiver. With these modifications, the TCP sender will recover from corruption losses without any reduction in congestion window. There is also a need to develop a unified analytical approach for the evaluation of TCP performance. We develop an analytical approach for the performance evaluation of NewRenoEln scheme. We compare the throughput results obtained by analytical calculations with results obtained by simulation and find them to be very close to each other. We also compare the performance of the proposed scheme NewRenoEln and the standard NewReno TCP via simulation as well as analytical approach, and find considerable improvement in throughput over wireless links.

The main task of this thesis was to explore the requirements of different telecommunications services on the network parameter, that are packet delay, packet loss, throughput, jitter. Make to test network service with WAN network emulator and load simulators. Next focus was to explore TCP congestion avoidance algorithms and focus on their throughput. The purpose of the first chapter is to introduce Quality of Service (QoS) for VoIP, video conversation, streaming audio, streaming video and data. The second chapter are to explore various types of delay, that are created from the sender through network device, transmission media to the recipient. The third chapter is to introduce NetDisturb application. It is application, that can emulate real properties of the data network and incoming traffic affect example packet delay, packet loss, bandwidth limit and so on. The key objectives of the fourth and fifth part is TCP protocol and TCP congestion avoidance algorithms. Hier are to introduce TCP protocol basic properties, congestion avoidance mechanisms and analysis of TCP congestion avoidance algorithms. In the practical part are created two laboratory exercises. First laboratory exercise is designed on the implementation QoS for various types of traffic. Second laboratory exercise is focused on the analysis TCP congestion avoidance algorithms and their throughput.

Many commercial devices now being produced have the ability to be remotely monitored and controlled. This thesis aims to develop a generic platform that can easily be extended to interface with many different kinds of devices for remote monitoring and control via a TCP/IP connection. The deployment will be concentrated on Medical devices but can be extended to all serial device interfaces. The hardware to be used in the development of this platform is an ARM Cortex M3 based Micro-Controller board which has to be designed to meet the requirement set by the Precept Health the founder of this platform. The design was conducted at Massey University in collaboration with senior engineer from the company. The main task in achieving the aim was the development of the necessary software layers to implement remote monitoring and control. The eCosCentric real-time embedded operating system was used to form a generic base for developing applications to monitor and control specific devices. The majority of the work involved in this project was the deployment of the operating system to the Micro-Controller. During the development process, several hardware issues were discovered with the Ethernet interface and were corrected. Using the generic platform, an application was developed to allow the reading of Bi-Directional pass through a communication protocol from 4 isolated serial input channels, to an Ethernet channel using TCP protocol.

[pt] Este trabalho é um estudo sobre o desempenho da arquitetura Diffserv proposta para permitir oferecimento de serviços com QoS na Internet. A ênfase do estudo é no desempenho de fluxos que utilizam o protocolo de transporte TCP.Inicialmente são revistos conceitos básicos, características do protocolo TCP em suas diferentes versões e os principais mecanismos de controle de tráfego. É feita, então, uma modelagem da arquitetura Diffserv visando a avaliação de seu desempenho. Procurou-se resumir os principais resultados sobre desempenho desta arquitetura disponíveis na Literatura e estabelecer um modelo de referência que permitisse a realização de avaliações específicas. Este modelo de oferecimento de serviços com QoS por um provedor é avaliado através do programa de simulação NS. Através de diversos casos analisados são feitas comparações de desempenho de fluxos TCP em diferentes classes de serviço,levando em conta o tipo de implementação e os mecanismos de encaminhamento de pacotes. / [en] This work presents a Diffserv architecture study to provide QoS in the Internet. The emphasis of the study is on the performance of flows that use the TCP transport protocol. Initially, basic concepts, TCP characteristics in its different versions and the main mechanisms of traffic control are reviewed. A modeling of the Diffserv architecture, aiming at the performance evaluation, is made. The main results on performance of this architecture in the available Literature were summarized and a reference model that allowed the accomplishment of specific evaluations was proposed. This model of QoS offering for a provider is evaluated through the NS simulation program. Through several analyzed cases, comparisons performance of TCP flows are made in different classes of service, talking into account the type of TCP implementation and the mechanisms of packet forwarding.

Τα σύγχρονα τηλεπικοινωνιακά δίκτυα οπτικών ινών χρησιμοποιούν την τεχνολογία WDM Wavelength Division Multiplexing) η οποία έχει κάνει εφικτή την αξιοποίηση – ως ένα βαθμό– του τεράστιου εύρους ζώνης της οπτικής ίνας. Στα πλαίσια της παρούσας διπλωματικής εργασίας θα παρουσιαστούν οι τρεις βασικές οπτικές αρχιτεκτονικές μεταγωγής (οπτική μεταγωγή κυκλώματος –OCS–, οπτική μεταγωγή πακέτου –OPS–, οπτική μεταγωγή ριπής–OBS–) οι οποίες μετατρέπουν τη διαθέσιμη χωρητικότητα σε ωφέλιμο throughput. Ιδιαίτερη έμφαση θα δοθεί στην αρχιτεκτονική OBS, η οποία έχει τραβήξει το ερευνητικό ενδιαφέρον τα τελευταία χρόνια, σαν μια ενδιαφέρουσα εναλλακτική της (ώριμης πλέον) αρχιτεκτονικής OCS. Συγκεκριμένα, θα διερευνηθεί η μετάδοση του TCP πρωτοκόλλου πάνω από OBS δίκτυα μέσα από λεπτομερείς προσομοιώσεις, και θα προταθούν κατάλληλες βελτιώσεις της αρχιτεκτονικής OBS. Ακόμα, θα περιγραφεί μια πρωτότυπη υβριδική αρχιτεκτονική οπτικής μεταγωγής ριπής. / Internet traffic has faced an exploding growth in recent years. The ever-growing demand for multimedia web services, as well as the advent of P2P technology, are driving core networks to their limits. This calls for the design of high capacity core networks, being able to serve the user’s high bandwidth requests. Optical networks have become a key part of the solution, mainly due to the vast capacity of optical fibers. Specifically, the advent of WDM technology has resulted in transmission capacities that have increased manifold in recent years. It is the router/switch throughput, however, that really transforms the raw bit rates into effective bandwidth. In this diploma thesis, we study the three basic optical architectures, that is Optical Circuit Switching (OCS), Optical Packet Switching (OPS) and Optical Burst Switching (OPS). Emphasis is given on OBS architecture, which has drawn research interest in recent year, as a possible replacement for the well-established OCS architecture. Specifically, we will study the transmission of TCP traffic over OBS networks through simulation, and propose modifications for the OBS architecture. Finally, a novel hybrid switch architecture will be proposed, combining the merits of OBS and OCS.

This diploma thesis deals with the design of temperature sensor and display units, which communicate via a WLAN network. The theoretical part provides an overview of various methods of temperature measurement including several specific types of temperature sensors, and also introduces the WLAN standard as well as TCP and UDP protocols. The design solution is focuses first on the system architecture and design of the communication protocol, followed by the implementation of both types of equipment, starting with selection of components, through design diagrams and printed circuit board patterns, up to production and commissioning. The following parts of this thesis deal with the design of programs for the controlling microcontroller units of both units and the configuration application for PC.

Depuis quelques années le trafic de l'internet mobile ne cesse d'augmenter. Cette croissance soutenue est liée à plusieurs facteurs, parmi lesquels l'évolution des terminaux, la grande diversité des services et des applications disponibles et le déploiement des nouvelles technologies d'accès radio mobile (3G/4G). À cet égard, le standard 3GPP pour les réseaux LTE propose une architecture offrant une gestion fine de la QoS (par flux). Ce modèle, hérité des réseaux mobiles traditionnels orientés connexion, soulève des problèmes en termes de scalabilité, efficacité et performances. Les travaux entrepris dans cette thèse ont pour objectif principal de proposer des solutions plus simples et moins coûteuses pour la gestion de la QoS dans les réseaux mobiles. À cette fin, à l'issue d'une étude et de l'évaluation de l'impact de la signalisation associée au modèle de QoS standard, deux modèles alternatifs ont été proposés. Nous proposons tout d'abord un modèle basée sur les mécanismes IP inspiré de l'approche DiffServ (par agrégat) largement étudié dans les réseaux IP fixes. Ce modèle fournit une gestion de la QoS simple, efficiente et rentable, tout en garantissant des performances équivalentes au modèle standard. Cependant, elle nécessite une remise à niveau de tous les eNB, et donc une longue phase de transition. En conséquence, nous proposons SloMo qui vise à améliorer l'expérience des clients mobiles, mais avec un objectif de déploiement plus rapide. SloMo est une solution de gestion implicite de la QoS depuis un point unique situé sur le chemin des communications. SloMo exploite la dynamique instaurée par le mécanisme de contrôle de flux de TCP. Il vise à recréer un goulot d'étranglement dynamique dans un équipement contrôlé par l'opérateur lorsque les points de congestion réels ne sont pas accessibles. Une fois ce goulot d'étranglement déporté, il est alors aisé d'effectuer une gestion de la qualité IP classique dans l'équipement supportant Slo-Mo. / The mobile data landscape is changing rapidly and mobile operators are today facing the daunting challenge of providing cheap and valuable services to ever more demanding customers. As a consequence, cost reduction is actively sought by operators as well as Quality of Service (QoS) preservation. Current 3GPP standards for LTE/EPC networks offer a fine tuning QoS (per-flow level), which inherits many characteristics of legacy telco networks. In spite of its good performance, such a QoS model reveals costly and cumbersome and finally, it remains very rarely deployed, thereby giving way to basic best-effort hegemony. This thesis aims at improving QoS in mobile networks through cost-effective solutions; To this end, after an evaluation of the impact and cost of signaling associated with the standard QoS model, alternative schemes are proposed, such as the IP-centric QoS model (per aggregate) inspired from the DiffServ approach widely used in fixed IP networks. This model provides a simple, efficient and cost-effective IP level QoS management with a performance level similar to standardized solutions. However, as it requires enhancements in the eNB, this scheme cannot be expected in mobile networks before a rather long time.Thus, we introduce Slo-Mo, which is a lightweight implicit mechanism for managing QoS from a distant point when the congestion point (e.g. eNB) is not able to do it. Slo-Mo creates a self-adaptive bottleneck which adjusts dynamically to the available resources taking advantage of TCP native flow control. Straightforward QoS management at IP level is then performed in the Slo-Mo node, leading to enhanced customer experience at a marginal cost and short term.

Le streaming adaptatif sur HTTP, désigné par HAS, est une technique de streaming vidéo largement déployée sur Internet. Elle utilise TCP comme protocole de transport. Elle consiste à segmenter la vidéo stockée sur un serveur web en petits segments indépendants de même durée de lecture et transcodés à plusieurs niveaux de qualité, désignés par "chunks". Le player, du côté du client HAS, demande périodiquement un nouveau chunk. Il sélectionne le niveau de qualité en se basant sur l’estimation de la bande passante du/des chunk(s) précédent(s). Étant donné que chaque client HAS est situé au sein d’un réseau d’accès, notre étude traite un cas fréquent dans l’usage quotidien: lorsque plusieurs clients partagent le même lien présentant un goulot d’étrangement et se trouvent en compétition sur la bande passante. Dans ce cas, on signale une dégradation de la qualité d’expérience (QoE) des utilisateurs de HAS et de la qualité de service (QoS) du réseau d’accès. Ainsi, l’objectif de cette thèse est d’optimiser le protocole TCP pour résoudre ces dégradations de QoE et QoS. Notre première contribution propose une méthode de bridage du débit HAS au niveau de la passerelle. Cette méthode est désignée par "Receive Window Tuning Method" (RWTM): elle utilise le principe de contrôle de flux de TCP et l’estimation passive du RTT au niveau de la passerelle. Nous avons comparé les performances de RWTM avec une méthode récente implémentée à la passerelle qui utilise une discipline particulière de gestion de la file d’attente, qui est désignée par "Hierarchical Token Bucket shaping Method" (HTBM). Les résultats d’évaluations indiquent que RWTM offre une meilleure QoE et une meilleure QoS de réseau d’accès que HTBM. Notre deuxième contribution consiste à mener une étude comparative combinant deux méthodes de bridages, RWTM et HTBM, avec quatre variantes TCP largement déployées, NewReno, Vegas, Illinois et Cubic. Les résultats d'évaluations montrent une discordance importante entre les performances des différentes combinaisons. De plus, la combinaison qui améliore les performances dans la majorité des scénarios étudiés est celle de RWTM avec Illinois. En outre, une mise à jour efficace de la valeur du paramètre "Slow Start Threshold", sthresh, peut accélérer la vitesse de convergence du player vers la qualité optimale. Notre troisième contribution propose une nouvelle variante de TCP adaptée aux flux HAS, qu’on désigne par TcpHas; c’est un algorithme de contrôle de congestion de TCP adapté aux spécifications de HAS. TcpHas estime le niveau de la qualité optimale du flux HAS en se basant sur l’estimation de la bande passante de bout en bout. Ensuite, TcpHas applique un bridage au trafic HAS en fonction du débit d’encodage du niveau de qualité estimé. TcpHas met à jour ssthresh pour accélérer la vitesse de convergence. Une étude comparative a été réalisée avec la variante Westwood+. Les résultats d’évaluations montrent que TcpHas est plus performant que Westwood+. / HTTP adaptive streaming (HAS) is a streaming video technique widely used over the Internet. It employs Transmission Control Protocol (TCP) as transport protocol and it splits the original video inside the server into segments of same duration, called "chunks", that are transcoded into multiple quality levels. The HAS player, on the client side, requests for one chunk each chunk duration and it commonly selects the quality level based on the estimated bandwidth of the previous chunk(s). Given that the HAS clients are located inside access networks, our investigation involves several HAS clients sharing the same bottleneck link and competing for bandwidth. Here, a degradation of both Quality of Experience (QoE) of HAS users and Quality of Service (QoS) of the access network are often recorded. The objective of this thesis is to optimize the TCP protocol in order to solve both QoE and QoS degradations. Our first contribution consists of proposing a gateway-based shaping method, that we called Receive Window Tuning Method (RWTM); it employs the TCP flow control and passive round trip time estimation on the gateway side. We compared the performances of RWTM with another gateway-based shaping method that is based on queuing discipline, called Hierarchical Token Bucket shaping Method (HTBM). The results of evaluation indicate that RWTM outperforms HTBM not only in terms of QoE of HAS but also in terms of QoS of access network by reducing the queuing delay and significantly reducing packet drop rate at the bottleneck.Our second contribution consists of a comparative evaluation when combining two shaping methods, RWTM and HTBM, and four very common TCP variants, NewReno, Vegas, Illinois and Cubic. The results show that there is a significant discordance in performance between combinations. Furthermore, the best combination that improves performances in the majority of scenarios is when combining Illinois variant with RWTM. In addition, the results reveal the importance of an efficient updating of the slow start threshold value, ssthresh, to accelerate the convergence toward the best feasible quality level. Our third contribution consists of proposing a novel HAS-based TCP variant, that we called TcpHas; it is a TCP congestion control algorithm that takes into consideration the specifications of HAS flow. Besides, it estimates the optimal quality level of its corresponding HAS flow based on end-to-end bandwidth estimation. Then, it permanently performs HAS traffic shaping based on the encoding rate of the estimated level. It also updates ssthresh to accelerate convergence speed. A comparative performance evaluation of TcpHas with a recent and well-known TCP variant that employs adaptive decrease mechanism, called Westwood+, was performed. Results indicated that TcpHas largely outperforms Westwood+; it offers better quality level stability on the optimal quality level, it dramatically reduces the packet drop rate and it generates lower queuing delay.

Content of master’s thesis is description field of Sliding window and it’s expansion algorithms, witch are Slow start, Congestion avoidance, Fast Retransmit and Fast Recovery algorithm. Thereinafter is described creation of model in Opnet Modeler’s simulation area. In this simulation area was analyzed reactions of average transfer speed onto variance of data size, lost ratio, latency in short and long time slot and variance of receiver’s buffer size. In last section of this document is method design witch makes it possible of transfer speed control through the use of receiver’s buffer size dynamic setting.